1. Field of the Invention
The present invention relates to telecommunications technologies, and more particularly to a method and system for supporting multiple levels of priority for voice or other types of traffic over a telecommunications network.
2. Background of the Art
Live voice communication is carried out over the Internet, and may thus be carried out over other wide area networks, local area networks, or other distributed networks, pursuant to Voice over Internet Protocol, or “VoIP.” In using VoIP, a person's speech is digitized and transmitted in Internet Protocol (“IP”) packets from one VoIP phone to a second VoIP phone. This may be implemented as a regular phone attached to an adaptor that converts analog signal to VoIP. The caller's speech to the recipient is transmitted in one stream of packets, and the recipient's speech is transmitted to the caller in a second stream of packets. Similarly, call setup and teardown control signals are sent and received in IP packets.
Currently, VoIP speech and call control packets are transmitted in accordance with the User Datagram Protocol (UDP), a communications protocol that offers a limited amount of service when messages are exchanged between computers in a network that uses IP, but that does not provide sequencing of the packets that the data arrives in. UDP sends a packet from one IP address to a destination IP address and assumes the packet will arrive at the destination. UDP does not have guaranteed delivery of packets as the more commonly known Transmission Control Protocol (TCP).
Under VoIP, voice packets between the caller and recipient are forwarded in the network by routers. A router may drop a packet or introduce uneven delays due to network congestion. Routers do not automatically re-route traffic to avoid congestion. As a result, a network router can drop call control or voice packets in VoIP phone calls. The effects of dropped VoIP packets can prevent the establishment of VoIP calls or cause garbled speech being delivered to one of the parties in a phone call.
To avoid such congestion and to enable live voice communication to be successfully carried out over a wide area network, VoIP packets may be processed in a separate priority queue that the routers process first. Differentiated Services (also referred to as “DiffServ” or “DS”) offers a protocol that specifies and controls network traffic so that certain types of traffic get precedence. For example, in a network transmitting both voice and other data, voice data (which requires continuous, uninterrupted transmission) may take precedence over other kinds of data that are not sensitive to temporal interruptions in their transmission. Such other types of data packets (e.g., ftp, http, video, etc.) are thus processed in a separate queue commonly referred to as the best effort queue. In order to provide the voice data packets with such differentiated service, the IP header of each voice data packet contains a Type of Service (ToS) byte, a portion of which is used to store a Differentiated Services Code Point (DSCP), a six bit value that specifies a particular “per hop behavior” (PHB) for the data packet, i.e., a predefined forwarding behavior for the data packet.
One such PHB has come to be known as the expedited forwarding per hop behavior (EF PHB), as described in Davie, B. et al., “An Expedited Forwarding PHB (Per-Hop Behavior)”, RFC 3246 (The Internet Society, March 2002), which is incorporated herein in its entirety by reference thereto. The EF PHB provides low loss, low delay, and low jitter VoIP services. The intent of the EF PHB is to provide a PHB in which packets marked for EF PHB ordinarily encounter short or empty queues, and maintain packet loss at a minimum by keeping the queues short relative to the available buffer space. Unfortunately, however, the EF PHB defines only one level of precedence and does not support multiple levels of precedence within the PHB. Thus, there currently exists a need for a multiple precedence level scheme to be applied to the EF PHB.
For example, during an emergency, such as a terrorist attack, severe weather, or other large-scale social calamity, there may be many people trying to make phone calls. With a VoIP telephone system, the increased volume of voice packets could exceed the bandwidth allocated by a router for VoIP. The result can lead to people being unable to place phone calls, or degrade existing calls to the extent that speech becomes unintelligible. If first-responders to an emergency are unable to establish critical communications, the results can be loss of life or significant loss of property.
As the commercial telephone sector is migrating toward a Packet Switching model rather than the Circuit Switching voice infrastructure that has been in use for many decades, the need is urgent to provide a VoIP system that will allow for such precedence based discrimination for voice traffic that is to be subjected to expedited forwarding.